similar to: asterisk as non root

Hi. Here is the output of the command root at pbx: ~ $ find / -name asterisk -exec ls -ld '{}' \; drwxr-xr-x 3 root root 4096 Apr 19 17:32 /usr/include/asterisk drwxr-x--- 3 asterisk asterisk 4096 Apr 19 17:32 /usr/lib/asterisk -rwxr-xr-x 1 root root 9719880 Apr 19 17:27 /usr/src/asterisk-11.25.1/main/asterisk drwxrwxr-x 3 1013 users 4096 Apr 19 16:56

root at PBX: /var/www/html $ /etc/init.d/asterisk start [ ok ] Starting asterisk (via systemctl): asterisk.service. root at PBX: /var/www/html $ ps aux | grep asterisk asterisk 1007 0.7 2.3 67128 23748 ? Ssl Apr19 8:49 /usr/sbin/asterisk -U asterisk -G asterisk root 4186 0.0 0.1 4192 1992 pts/0 S+ 17:30 0:00 grep asterisk root at PBX: /var/www/html $

Hi. thanks a lot for your replies. I did stop the services and i did issued the the "chown" and "chmod" commands listed in the guide. It is necessary to compile it, instead if using the apt-get version What am i missing? On Wed, Apr 19, 2017 at 10:47 PM, Antony Stone < Antony.Stone at asterisk.open.source.it> wrote: > On Wednesday 19 April 2017 at 18:48:29, Atux

hi. i would like to be able to reboot the system from my extension. is that possible? if yes, how? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20170406/55c7bee2/attachment.html>

I just updated to 11.7.0 When listing channels from command line I get: Channel Location State Application(Data) 0 active channels 0 active calls 0 calls processed Privilege escalation protection disabled! See https://wiki.asterisk.org/wiki/x/1gKfAQ for more details. Conference Bridge Name Users Marked Locked? ================================ ====== ======

Hi all again... I did a chroot patch with ability to change uid and gid. I dunno if you all approve with the current implementation but the patch is small and it works here. remember to CC, since i'm not on this ml... //Ian Kumlien PS. To anyone on vorbis-dev, I'm *REALLY* sorry must be tired or something.. =) DS. <p><p><p> -------------- next part -------------- A

Hi. i have an asterisk 11 installation that i run in my soho environment. My system has mysql to store all the cdrs. I would like make use of the mysql and store numbers and names. eg +4922123456789 "Atux Null". So when the +4922123456789 calls in my system the name "Atux Null" will pop up next to the number. at the moment i have a database called MyNames in mysql that has this

so any ideas, please? On Tue, Apr 10, 2018 at 1:46 PM, Atux Atux <atuxnull at gmail.com> wrote: > after adding the ww: > root at Pbx: /etc/asterisk $ asterisk -rvvv > Asterisk 11.25.3, Copyright (C) 1999 - 2013 D == Using SIP RTP TOS bits > 184 > == Using SIP RTP CoS mark 5 -- Executing > [9211123456 at AllCalls:1] Goto("SIP/500-00000003",

No idea on how to write it in my system. On Thu, Jan 11, 2018 at 12:17 AM, John Kiniston <johnkiniston at gmail.com> wrote: > There's some example code in the Dial-Users context of the basic-pbx > samples that might be of use in implementing it. > > They are checking a DEVICE_STATE to see if a phone is BUSY, You could > change it to be a database call or implement custom

Thanks a lot for the reply. I did follow that already, but i do have a problem. Here is my extensions.conf part for that particular number exten => 6912345678,1,Answer() exten => 6912345678,n,MYSQL(Connect connid 127.0.0.1 root mypasswd asterisk) exten => 6912345678,n,MYSQL(Query resultid ${connid} SET NAMES utf8) exten => 6912345678,n,GotoIf($["${connid}" =

Hi. thanks a lot for your reply. i will download the newer libical software. Could you elaborate on icalendar with google calendar config and calendar.conf, please? On Thu, Mar 15, 2018 at 3:00 PM, Ludovic Gasc <gmludo at gmail.com> wrote: > I never use caldav mode, always icalendar with Google Calendar. > > BTW, you use old versions of libical, Asterisk and Debian, I recommend

Hi. Thanks for the idea for calendar, it sounds better. i did not manage to make it work though. i am running debian 8 32 bit with asterisk 11.25.3. I have installed the packages libneon27-dev & libical-dev then in /etc/asterisk the file calendar.conf has the following entries: [Gcalendar] type=caldav url=https://www.google.com/calendar/dav/atuxnull at gmail.com/events/ user=atuxnull at

Hi, we started using centos7 to run kvm+libvirt. For novnc to virtual machines I like to start websockify as a daemon by libvirt (as a hook) Under Debian I used the line: # start-stop-daemon --start --chuid $RUNUSER --make-pidfile --pidfile $PIDFILE --startas $PROGRAM -- $OPTIONS which creates a nice pidfile. In Centos I figured out, to include /etc/init.d/functions and I tried: # daemon

Hi. in my system i have a conference room where someone can call it eg 698 dial the PIN eg 1234 and enter the room as a user. The admin enters in through a different number and PIN. I would like to have a call file and call all participants eg 610-619 at certain time of the day and give them access to the conference. During my try i managed to create a call file where it calls the a SIP phone and

That is the general idea. But how do i make it work? is there somewhere ready? On Wed, Jan 10, 2018 at 6:39 PM, John Kiniston <johnkiniston at gmail.com> wrote: > Define your *72 and *73 extensions in your internal context, Have them set > a value in the ASTDB that you then check when dialing your handsets. > > The same can be done for call forwarding, store a number in the

Hi. I have an Asterisk Server (A) where it acts as the main gateway to offer services. There are different asterisk servers (B -D) that connect as extensions to the Server A. I would like to implement TLS and SRTP for these extensions, but have the non secure as well for other extensions. for example the extensions 4500-4504 be with TLS/SRTP and the rest be non secure(ordinary). Is there a guide

Hi. I am running asterisk 11 and i have usb 3g dongles to make my gsm calls with the following config in extensions.conf exten => _9X.,1,Dial(Dongle/dongle800/${EXTEN:1},120,KT) exten => _9X.,n,Hangup(${HANGUPCAUSE}) By dialing 9 it opens the dongle to make a call. I would like to restrict my caller id. so when i place a call from this dongle, it will send on the other end *blocked number*

Hi. in my home office i operate my asterisk and have an IVR that has the business hours 9-5 and everytime i edit it to load the bank holidays (New Years eve, christmas, easter, whatever else). I would like to be able to load in the Asterisk's DB or in a file for all the year or years the planned holidays. Then it will be read from that file to operate accordingly. Is there a hint on how to run

hi. i am running asterisk 11 and i am stuck with the feature codes. how do i setup them. Now the system has. PBX*CLI> features show Builtin Feature Default Current --------------- ------- ------- Pickup *8 *8 Blind Transfer # # Attended Transfer One Touch Monitor Disconnect Call * * Park Call One Touch MixMonitor Dynamic Feature Default Current --------------- ------- ------- (none) Feature

hi. currently i am running the phonebook in astdb with *database put cidname 0123456789 "name_surname"* and i retrive it with *exten =>9876543210,1,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})})* Now, my system has mysql and i got all my contacts in there in a database is called *asterisk *and a table called *addressbook**. *password of the mysql is *whateverpasswd* how do i